Audio data can be sampled at a variety of different bit rates. The sampling rate defines a number of samples per second taken from a continuous signal to make a discrete signal. One approach is to sample the audio data by using linear interpolation. Linear interpolation is a simple and fast form of interpolation, but the quality of the sampled audio can be poor if the difference between the sampling rates of the original audio data and the outputted data is large. For example, if the audio data is originally sampled at 16 KHz and a computing device is configured to sample and to output the audio data at 44.1 KHz, using linear interpolation to sample the 16 KHz audio data to 44.1 KHz would introduce a large amount of aliased signals at higher frequencies and therefore, would result in a poor quality output. Another approach is to sample the data using a more complicated band-limited interpolation. Band-limited interpolation yields good interpolation performance, but the complex calculations associated with band-limited interpolation are time consuming when compared to linear interpolation and requires a large logic circuit.
In view of the foregoing, there is a need for methods and circuitries that could quickly sample audio data and have a small hardware footprint.